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minyue@webrtc.org 2c0cdbce22 Estimating NTP time with a given RTT. 
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.

This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.

An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.

BUG=

TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:52:43 +00:00
..
base
Disable TestDTLSConnectWithSmallMtu on all platforms.
2014-10-09 07:52:03 +00:00
build
Use openmax_dl on all ARM (v7 or higher) platforms.
2014-10-09 04:13:02 +00:00
common_audio
common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
2014-10-09 08:47:02 +00:00
common_video
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle/xmllite
Mac: adds missing _DEBUG flag to mac debug builds.
2014-10-06 22:04:11 +00:00
modules
Estimating NTP time with a given RTT.
2014-10-09 10:52:43 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Import LappedTransform and friends.
2014-10-01 17:42:18 +00:00
test
GN: Add common configs to tools and test.
2014-09-30 19:07:58 +00:00
tools
Check on the existence of report directory
2014-10-06 17:21:27 +00:00
video
Wire up CPU adaptation in WebRtcVideoEngine2.
2014-10-03 11:25:45 +00:00
video_engine
Estimating NTP time with a given RTT.
2014-10-09 10:52:43 +00:00
voice_engine
Estimating NTP time with a given RTT.
2014-10-09 10:52:43 +00:00
.gitignore
…
BUILD.gn
GN: Enable libvpx, add link target and convert some test targets
2014-09-30 18:05:02 +00:00
call.h
Wire up CPU adaptation in WebRtcVideoEngine2.
2014-10-03 11:25:45 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
2014-09-30 14:21:10 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VP8/H264 defaults through video_encoder.h.
2014-09-18 12:42:28 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
webrtc_examples.gyp
Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
2014-09-27 18:10:30 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Moves xmllite's unittests to rtc_unittest.
2014-10-02 18:43:47 +00:00
webrtc.gyp
Moves xmllite's unittests to rtc_unittest.
2014-10-02 18:43:47 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.